my DAC has the TPA6... (? can't remember) chip used as both headphone amp and line driver. current feedback, very fast and all but when I first received it was outputting a weird noise. returned it and manufacturer fixed it but I think I got to know the drawbacks of a very fast amp.

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we all love a good ol' stereotype until it's against us

There are, I think, basically two approaches to slew rate, or rather how it should be done.

One was, in its time, advocated by Sansui, Kenwood/Trio and Pioneer, to mention a few, and basically it boiled down to the idea - go for it, you can't have too much.

I disliked, and still do, such an approach by default. Just because you can doesn't mean you always should. Ultimately, one could say it darws the designers' minds away from other, possibly more important, aspects of amplifier design. Consequently, most of their products, including higher price range ones, did have wild slew rates, but relatively poor load tolerance. In my view, you have incomparably better chances of meeting a loudspeaker with an evil impedance modulus than any signal needing 250 V/uS voltage slew rate.

It seems to me they got wild voltage slew rates at the express expense of current slew rate and overall loda tolerance.

Then there is the other, cautious school, well exmplified by say Studer/reVox and Sony. They would first get an "internal" slew rate of say 300+ V/uS, but then they would install an input filter typically with a -3 dB cutoff point of 200 kHz. Eventually, they would decalre both values, and it was up to customer to make up his/her mind whether they could live with it.

I have seen Studer/reVox specs like this: Absolute slew rate 120 V/us, effective slew rate (with input filter) 80 V/uS. I have seen Sony specs claiming an "internal" slew rate of 350 V/uS, but with input filter down to less than 150 V/uS.

I would now draw your attention to Nigel's post in which he describes the UK view of curtailed bandwidth. There is logic in that as well, I mean, what's an audio power amp doing at say 500 kHz? The very fact that some of them have been pushed to and above 1 MHz makes no sense to me, even when I take into account excpetionall low phase shifts at 20 kHz.

Obviously, I agree with Studer/reVox and Sony logic - extend it as far as it costs you little effort to get to, in my case between 300 and 400 kHz (closed loop, 28.3 Vrms nominal at 1 kHz, measured at the -3 dB point (i.e. where it delivers only 20 Vrms). Then make sure the input filter is in place, strctly first order low phase shift, with a cutoff point of around 200 kHz.

Lately, I've been wondering if even that is unnecessarily high, but I figure that -3 dB at 200 kHz is around -1dB at 100 kHz, -0.5 dB at 50 kHz and -0.25 dB at 25 kHz, and I find that an acceptable loss.

my DAC has the TPA6... (? can't remember) chip used as both headphone amp and line driver. current feedback, very fast and all but when I first received it was outputting a weird noise. returned it and manufacturer fixed it but I think I got to know the drawbacks of a very fast amp.

Some years ago, a design of mine was giving me trouble, it tended to be not completely stable. I spent a lot of time mulling over it, but once it gave some "birdies", I knew where to look. So, instead of the usual 220 nF cap doing all the decoupling, I lined 'em up - 100uF in parallel with 1 uF metallized foil (Wima) in parallel with 220 nF (metallized foil, Wima).

And lo and behold, no more "birdies", no untoward behavior at all. When one gets his fingers and ears burned personally, one learns ever so quickly.

I think to be able to offer as much as you can seems ideal . Like camera filtering it should be available . That's not to turn everything red . Just a subtle filter . As people said for cameras , it filters UV , it gives a very slightly warm effect , it protect the lens . Sometimes I took mine off for special shots .

I am being very lazy . I have to use an LM317 for the first time in years . It will be cheaper than using a transistor and less components as a constant current source . From memory the voltage limit is in to out rather than absolute ? I intend to use 49 V in 24 V out at 42 mA . I don't intend to buy the high voltage version . I think that's OK ? If not BD140 will do fine . The price is about the same . LM 317 doesn't quote dissipation as such ( 50 + 5 C / W + ambient < 125C ) , looking at it 1.5 W seems about right ( it will shut down if too hot ) .

Nigel, you have not thought Rate of Change through. It does NOT take much high frequency energy to give a high dV/dT to an audio signal, JUST ALLOWED BANDWIDTH.

Hi John . I 99% accept that . Just want as much debate as possible . I sense some good stuff is coming out . Any fool can see high slew rates are good , can we make them work even better ? I have this little doubt that very large bandwidth helps anyone . Perhaps 30 kHz - 1 dB is as much as we need ? I do accept that less than that sounds wrong . I have no idea why because most speakers are much worse . It is not frequency response so much as mixing non music into the signal ?

It is said valves are less fussy about this . Hard to say because keeping it going above 20 kHz isn't always easy with valves . They are not as good on PSRR so it is not a given .

One thing Wave said before is symmetrical slewing if we can . I don't agree it is paramount , however it must be a good thing if we can . I see asymmetrical slewing a bit like riding a bicycle with one leg .

One thing Wave said before is symmetrical slewing if we can . I don't agree it is paramount , however it must be a good thing if we can . I see asymmetrical slewing a bit like riding a bicycle with one leg .

I, on the other hand, believe that they are of paramount importance, and that this is what gives us anything from no to much ambience. Spatial detail.

I hit on the idea quite by chance; I wish I could say I was really smart and thought it out, but I didn't.

I was fooling around with a preamp based on op amps. It worked fine in terms of function, but its sound, while well articulated, was about as flat and 2D gets. Soundstage heigth and depth all equal to zero, just very good stereo separation. Whoever made it was my kinda guy, he used single op amps only, no duals. It was a Hungarian product, very decently made and I would say very well thought out overall.

I had just received a plastic rail full of Analog Devices AD818. So I thought, heck, why not? I pulled out the TLO op amps (thank God, they were in quality gold plated Amphenol sockets, so it was easy) and installed the AD op amps. Got lucky, the whole thing worked right off the bat.

The difference was, and I use the word with prudence, stunning. As if it was a totally different product. Lots of soundstage detail in all aspects, the usic came alive.

My first thought was aha!, slewing rate at work, after all, the TLO71 series is hardly known for its speed, 10 V/uS, and the AD 818 is like 350 V/uS. Anyway, to cut a long story short, in the end, after more testing there and elsewhere, I realized that this was not due to a much higher slew rate (althought I'm sure that helped as well), but to a MUCH shorter settling time. TLO71 is rated at 2,000 nS for settling down to 0.1%, while the AD 818 is rated at 90 nS for settling down to 0.01% (note the added zero), or 45 nS for the same 0.1%. That's merely 44.5 TIMES faster. Practically no overhang.

I tried and retried the same test several times over later on, and each and every time, ambience was added upon changing the op amps if the new one had a short settling time, and being an AD man, ALL my op amps have short settling times (AD 818, 828, 829, except the venerable OP37, which is rated at 1,500 nS, but it's vintage by now).

Therefore, I am deeply convinced that Keith Johnson is dead right - we need to pay much more attention to as fast as possible settling time more than to wild rise times and slew rates. As ever, a good balance between all of them will probably yield the best results overall, but with three figure slew rates, I really have nothing left to do with those fast op amps.

The only additional benefit I could come up is to add a discrete current boost stage, consisting of two 1N4148 and say MPSA 56/06 or BC 639/640 transistors, or two BC 550/560 B trannies instead of the diodes. However, this is a step which simply never fails no matter which op amp one uses, and no matter what the manufacturer says it can do current wise.

Try this on an op amp based headphones amp and you WILL hear the difference. If you walk the whole nine yards, and use four diodes, a driver and an output pair proper (I suggest BD 139/140 for Nige, and MJE 15030/15031 for everybody else ), I think you will be surprised at the quality of sound you hear, not even to mention the available force, depth and control for the bass lines. You may even be surprised that your headphones can deliver that kind of bass lines.

And if you take the time and trouble to match the transistors propely, you will end up wondering what those offering headphones amps at crazy prices mean at all.

Again I like that . I found by accident that a push pull VAS seemed to be better . If you like a bicycle peddled with two legs which sources and sinks equally . One has to agree with Douglas Self and say if slewing is enough the asymmetry in itself should not be a problem . However the mechanism which causes the asymmetry may not be so nice . Mr Self has no time for the push pull VAS . As he says it will not be perfectly balanced . His argument seems to be it might need adjusting . In practice it doesn't once a design is finalized . Also it has a simplicity about it which solves many problems downstream . For example the current mirror on the second stage helps set the balance of the input pair . The first stage without any matching is usually within 2% which is good enough for anyone . Distortion is very low even at 50 kHz including the output devices . One day I will sling some extra FET's on the other side and make a very economical bridge amp just to see what I get . It might be useful to have the same load each side . The reason I think the push pull VAS works so well is that the transistors are emitter coupled . Thus some common base virtues . If that isn't two legged peddling I don't know what is ?

Nige, I think many of your dilemmas would be solved if you opted for a fully complementary topology, input to output. I am not going to say it sounds better per se, but I will say it can be made to sound better much easier than any other topology known to me.

It seems to be THE logical topology, given how it tracks the signal. It has been known for a long time that in fact very few signals are perfectly symmetrical sine waves, and a fully complementary topology is better suited for just that event, given that it has a separate half for plus, and another seoparate part for the minus part of the signal.

True, it tends to surpress even harmonics, and I have seen it actually accused for this fact by some dimwited dodo who it seems wanted lots of even harmonic distortion. Personally, I don't want much distortion of any kind, but I'm not a freak who expects THD figures like 0.001%. I have learnt long ago, in my own home, that it's possible for an amp to have a declared THD figure of 0.3% into 4 Ohms and still sound the pants off the vast majority of other far better specified products. With just 50/70 Wrms into 8/4 Ohms and a SEPP output stage.

And, as we all know, a great amplifier starts with a decision on topology.

I discovered that when designing my valve amp . The complimentary asymmetry can be used to reduce distortion considerably . Even more so using a mixture of pentode and triode if SE amp . I felt I was cheating when doing that . However it more than works well . There is a dogma with valves , triodes or nothing . Well that is a bit daft as distortion , gain and spectrum of distortion might be worse . We have no P devices with valves . Still we have opportunities to get a free lunch . I use transistor CCS with them which work well . I feel a transistor is ideal for that , it should have very little distortion and certainly no transistor sound ( supposed opaque quality ) . I then use a small amount of negative feedback more than anything to have a bit of damping factor . As the amp is already nice no nasty high order stuff walks in ( - 80 dB + 5 th harmonic ) .

Therefore, I am deeply convinced that Keith Johnson is dead right - we need to pay much more attention to as fast as possible settling time more than to wild rise times and slew rates. As ever, a good balance between all of them will probably yield the best results overall, but with three figure slew rates, I really have nothing left to do with those fast op amps.

funny you should mention Keith Johnson, I found about Spectral only recently. haven't listened to their amps but my gut tells me there must be something special about them.
concerning settling time it would make all the sense in the world. but there's one thing that negates that (again I have to agree with Mr. Curl wrt that aspect): oversampling filters in most DACs seem to do way more harm with their pre- and post-ringing. some filter types minimize those artifacts but still you can't eliminate them completely. my DAC is based on the WM8742 and it has user-selectable filters (5 of them). I was curious to look at the output of the DAC on a scope with square waves and none of the 5 filters looked nice in this respect. yes, I know, what one actually sees there is a band-limited square wave that can't possibly look any other way once it's band-limited. but what should be noted is that (correct me if I'm wrong) I think settling time-related artifacts in an amp are generally much less severe.

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we all love a good ol' stereotype until it's against us